[Asterisk-Users] Asterisk SIP Packet Time (20ms)

[Andres]

On Tuesday 23 December 2003 11:40, Rich Adamson wrote:
> There's no reassembly with udp, and there is no sense of packets arriving
> in the same order as what was sent. Udp is a best-effort low-overhead way
Right, UDP itself does not care about order, but at the application layer you 
can keep track of it.  You can design your application to buffer X packets 
and then reorder them according to sequence numbers.

> of transmitting data (with UDP often times referred to as the Unreliable
> Data Protocol). Changing to TCP would allow reassembly, however the
> overhead would be substantial.
>
> ------------------------
>
> > The problem occurs when the software is expecting the packet in a certain
> > timeframe so that it can reassemble it in a timely manner.  It's not a
> > big deal with a web page or something along that lines.  But when a voice
> > application cannot get reassembled in a timely manner, you'll surely
> > notice it!
> >
> > -----Original Message-----
> > From: Joel Maslak
> > To: asterisk-users at lists.digium.com
> > Sent: 12/23/2003 10:41 AM
> > Subject: Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
> >
> > On Tue, 23 Dec 2003, Rich Adamson wrote:
> > > If a collision or dropped packet occurs (in a voip udp environment)
> >
> > there
> >
> > > is no way to retransmit the missing/damaged packet. Missing one packet
> >
> > isn't
> >
> > > a big deal, but if you have collisions and/or dropped packets, there
> >
> > is a
> >
> > > very high probability that lots of packets will be dropped. If too
> >
> > many
> >
> > > are dropped, you'll hear the result in the undecoded voice as choppy
> > > voice.
> >
> > Actually, collisions occur at Layer 2, not Layer 3, and the layer 2
> > hardware automatically resends packets involved in a collision - layer 3
> > is never aware of it happening (although it may cause additional delay).
> > Eventually the ethernet card will give up if too many collisions occur
> > during retries, but this is very rare in practice unless the network is
> > *VERY* loaded.
> >
> > > Assuming alaw/ulaw codecs in use (about 80k bps), a half duplex 10 meg
> > > ethernet would handle roughly 20-25 rtp sessions before bumping into
> >
> > the
> >
> > > problem (your milage may vary). The majority of the folks on this list
> > > seem to be running home/soho systems and would likely never run into
> >
> > the
> >
> > > issue. But the heavier users will.
> >
> > For a duplex mismatch, my experience is that if one end on a 100 Mb/sec
> > link is half and the other is full, bandwidth is limited to about 8
> > Mb/sec
> > max.  This is based on some tests I've accidentally conducted.  If you
> > try
> > to send 9 Mb/sec over that link, yes, some packets will get dropped as
> > they simply won't fit.  (But I do agree that for a half-half link, you
> > can
> > get about 20 Mb/sec)
> >
> > --
> > Joel
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>
> ---------------End of Original Message-----------------
>
>
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